sprang@webrtc.org
54ae4ffb9e
Add callbacks for receive channel RTCP statistics.
...
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.
TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.
BUG=2235
R=henrika@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19 13:26:02 +00:00
andresp@webrtc.org
e682aa5077
Refactoring MediaOptimization so it can easily be turned into a thread-safe class.
...
BUG=2732
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5322 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19 10:59:48 +00:00
stefan@webrtc.org
faada6e604
Integrate fake_network_pipe into direct_transport.
...
TEST=trybots
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5321 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 20:28:25 +00:00
fbarchard@google.com
8f99a18119
Port scale and compare functions to pepper_33 and mips.
...
BUG=none
TEST=validator passes with new toolchain.
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5320 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 19:51:37 +00:00
kjellander@webrtc.org
5fe2d65c43
Remove metrics_unittests
...
This target has been merged into video_engine_tests in r5284.
BUG=webrtc:1843
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5319 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 13:27:37 +00:00
pbos@webrtc.org
8a54417968
Remove media_file from VideoEngine dependencies.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 10:00:29 +00:00
mflodman@webrtc.org
b429e516a9
cpplint cleaning new API and its implementation files.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6089005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5317 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 09:46:22 +00:00
mflodman@webrtc.org
bcd124cdba
Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand().
...
Follow up steps is to support NackConfig.rtp_hostory_ms and/or increase fake encoder bitrate.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5316 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 09:45:45 +00:00
mflodman@webrtc.org
1fa41be66a
Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5315 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 09:44:53 +00:00
sergeyu@chromium.org
8ae72560dd
Make MouseCursor mutable
...
MouseCursor objects were previous immutable which makes it harder to
implement deserializers when MouseCursor is sent over IPC in Chromium.
R=dcaiafa@chromium.org
Committed: https://code.google.com/p/webrtc/source/detail?r=5310
Review URL: https://webrtc-codereview.appspot.com/6059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5314 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 02:18:01 +00:00
fischman@webrtc.org
f8be8df33a
audio_processing_unittest: unbreak clang compilation.
...
BUG=2735
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5313 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 23:46:39 +00:00
fischman@webrtc.org
179908c81c
JNI Audio: remove dead members.
...
BUG=2735
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5312 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 23:46:14 +00:00
sergeyu@chromium.org
e4c927208b
Revert "Make MouseCursor mutable"
...
This reverts commit a6db8ab8bc4b569a26633b0ca3665297f1a5349b.
TBR=dcaiafa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/6079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5311 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 22:48:50 +00:00
sergeyu@chromium.org
8fd1d26536
Make MouseCursor mutable
...
MouseCursor objects were previous immutable which makes it harder to
implement deserializers when MouseCursor is sent over IPC in Chromium.
R=dcaiafa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/6059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5310 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 22:19:12 +00:00
fischman@webrtc.org
af320fd2f7
The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation.
...
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6019004
Patch from Rafael Lopez Diez <rafalopezdiez@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5309 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 21:33:27 +00:00
fbarchard@google.com
50f7b2da5d
roll libyuv to r915 for webview jpeg build fix and NaCL pepper_33 initial support.
...
BUG=none
TEST=try bots
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5308 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 18:18:17 +00:00
pbos@webrtc.org
052fa6243a
Stop transport in test SuspendBelowMinBitrate.
...
Avoids race when packets are still left in the network while the Call is
being destroyed.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/6009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5307 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 11:19:58 +00:00
mflodman@webrtc.org
e6b871bb29
Added method for getting default module state and protect agains a
...
read/write race for child_modules_.
BUG=2731
TEST=tsan
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5306 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 08:30:40 +00:00
fbarchard@google.com
9df6674b26
Scale down by 4x with box filter. Fix for 1 pixel wide bilinear filter. Fix for I420ToARGB overread on V plane that causes valgrind fail.
...
BUG=none
TESTED=gcl try libyuv_r911 --bot=linux_valgrind
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5305 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 08:25:31 +00:00
pbos@webrtc.org
eb7b7bce3d
Modify video_render/ to allow a single old frame.
...
This stabilizes tests as a single frame reaches end-to-end, as well as
allowing slow or heavily-loaded systems to see any video updates even if
the frame takes more than 500ms in the pipeline.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=2724
Review URL: https://webrtc-codereview.appspot.com/5949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5303 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 18:24:37 +00:00
fischman@webrtc.org
5b3c67ef25
objc/README: Remove outdated advice about target_os.
...
BUG=chromium:248168
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5979005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5302 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 17:15:19 +00:00
pbos@webrtc.org
919f87fb36
Delete capturers after destroying streams in test.
...
Since the renderers in CallTest.SendsAndReceiveStreams also stopped the
capturers they must be deleted after the VideoReceiveStream is stopped
or an use-after-free may occur.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5300 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:55:54 +00:00
asapersson@webrtc.org
e7b1e11283
Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."
...
> Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
>
> > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> >
> > R=holmer@google.com
> >
> > Review URL: https://webrtc-codereview.appspot.com/5049004
>
> TBR=asapersson@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5799004
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:40:36 +00:00
bjornv@webrtc.org
1e7d61270c
Simplification of histogram normalization in delay estimator.
...
- Replaces a for loop with a single element update to save complexity. No regression in performance seen on set of recordings.
- Removes UpdatesMadeUponChange() and put code straight into ProcessBinarySpectrum().
BUG=None
TESTED=module_unittest, trybots, verified manually on set of recordings.
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5298 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 13:37:28 +00:00
pbos@webrtc.org
5ab756703e
Revert r5294 to re-roll r5293.
...
To fix races in test each stream now owns its own encoder/decoder.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 12:24:44 +00:00
bjornv@webrtc.org
5c64508b03
Adds robust validation functionality to the delay estimator
...
Evaluated over a 51 recordings:
False positives went from 4.4% to 0.7%
Missed detections unchanged at 0.8%
No increase in complexity, but need to re-evaluate that.
TESTED=trybots, unittests, verified against Matlab implementation
BUG=None
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5296 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 10:57:53 +00:00
sprang@webrtc.org
87ad57bc75
Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver
...
The iterator is incremented both in loop header and loop body. Should
only be incremented in header.
BUG=2727
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5295 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 07:43:51 +00:00
turaj@webrtc.org
41e2615e02
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
...
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
>
> BUG=
> R=mflodman@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5409004
TBR=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-15 18:42:32 +00:00
solenberg@webrtc.org
341e91441a
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
...
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 23:57:54 +00:00
turaj@webrtc.org
e1bc6c8d8b
Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing.
...
BUG=
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5292 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 22:04:18 +00:00
stefan@webrtc.org
dd393e7b9d
Measure pacer queue size based on when packets are inserted rather than captured.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5291 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 22:03:27 +00:00
turaj@webrtc.org
167b6dfc73
Fix jitter buffer delay estimate.
...
BUG=b/12099925
R=niklas.enbom@webrtc.org , niklase@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5289 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 21:05:07 +00:00
wu@webrtc.org
24301a67c6
Update talk to 58174641 together with http://review.webrtc.org/4319005/ .
...
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 19:17:43 +00:00
mflodman@webrtc.org
92c2793154
Adding REMB to receive stream configuration, the send side will always
...
react to incoming REMB for now.
Adding a test to verify the receive side is generating RTCP REMB and
will follow up with a send side test as soon as the bitrate stats are
wired up for the new API.
TEST=See above.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5286 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 16:36:28 +00:00
asapersson@webrtc.org
86bb56a7f5
Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
...
> Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
>
> R=holmer@google.com
>
> Review URL: https://webrtc-codereview.appspot.com/5049004
TBR=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 16:16:45 +00:00
pbos@webrtc.org
0a222eba69
Merge metrics_unittests into video_engine_tests.
...
metrics_unittests will be removed as soon as trybots catch up with LKGR,
that way we don't have to break any tryjobs during.
BUG=1843
R=kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5284 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 14:31:47 +00:00
pbos@webrtc.org
1d096901ac
Move realtime tests to webrtc_perf_tests.
...
New binary not to be run on our VMs as they result in flaky tests. These
will instead be run on baremetal machines.
BUG=2710
R=kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5283 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 12:48:05 +00:00
mallinath@webrtc.org
62451dcba0
Update talk to 58157731.
...
R=wu@webrtc.org
TBR=wu@webrc.org
Review URL: https://webrtc-codereview.appspot.com/5339005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5282 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 12:29:34 +00:00
sprang@webrtc.org
6811b6e308
Callback for send bitrate estimates - new roll
...
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as
r5259, after which flakiness was detected and a rollback was performed
at r5261.
Patch Set 1 of this issue is the code submitted in r5259. Subsequent
patch sets fixes a race condition which caused the seen problems.
The root cause was a dead lock between a thread sending rtp packets and
and a timed module processing thread:
webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock
webrtc::Bitrate::Process() // Get Bitrate lock
webrtc::RTPSender::ProcessBitrate()
webrtc::ModuleRtpRtcpImpl::Process()
...
webrtc::Bitrate::Update() // Get Bitrate lock
webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock
webrtc::RTPSender::SendToNetwork()
...
This is fixed in Bitrate::Process() by releasing the lock before
calling the callback.
BUG=2235
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 09:46:59 +00:00
mflodman@webrtc.org
f3973e81d5
Make sure channels in the same call are in the same channel group.
...
Tested manually. I'll make a follow CL with a proper test once review.webrtc.org/5619004 has been committed.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5280 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 09:40:45 +00:00
henrik.lundin@webrtc.org
e9abd591d7
Making RemoteRateControl::min_configured_bit_rate_ configurable
...
The minimum bitrate can now be configured from WrappingBitrateEstimator.
BUG=2698
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5279 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 08:42:42 +00:00
wu@webrtc.org
a9890800e0
Update talk to 58127566 together with
...
https://webrtc-codereview.appspot.com/5309005/ .
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:21:03 +00:00
turaj@webrtc.org
a92baead39
ACM 2 compatibility with ACM 1.
...
Removing an unregisterd codec from ACM 1 does not result in an error, so should be for ACM 2. Also ACM 1 has post-decode VAD on and AMC 2 needs to have it on by default.
BUG=
Test=trybits
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5276 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:10:44 +00:00
wu@webrtc.org
2018269dc3
Revert 5274 "Update talk to 58113193 together with https://webrt ..."
...
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
>
> R=mallinath@webrtc.org , niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:54:25 +00:00
wu@webrtc.org
a129b6cd13
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
...
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:40:39 +00:00
henrike@webrtc.org
451745ec05
Complete rewrite of demo application.
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BUG=2122
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5273 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 16:55:37 +00:00
asapersson@webrtc.org
88ac63abc6
Remove overloaded CpuOveruseMeasure function.
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R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5272 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 14:37:33 +00:00
fischman@webrtc.org
df7b1d6e39
AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure.
...
Also silence a 'cd' that would otherwise emit the path/to/talk.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5271 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 22:36:22 +00:00
henrike@webrtc.org
9ee75e9c77
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
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BUG=N/A
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:42:44 +00:00
fischman@webrtc.org
f41f06b916
PeerConnection(java): rationalize pointer-to-jlong conversion.
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In r4665 I went a bit crazy with the manual reinterpretation of a pointer to a
jlong (to avoid undefined behavior) but that's what reinterpret_cast<> is for.
So use it directly now.
Added a do-nothing DataChannel to AppRTCDemo to regression test this, since the
only repro I've found of the original bug requires ARM ABI (PeerConnectionTest
on ia32 fails to repro).
BUG=2302
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5269 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:07:18 +00:00